1. Field
The present disclosure relates to a system and a method for controlling a Voice over Internet Protocol (VoIP) service and, more particularly, to a system and a method for controlling a VoIP service that enable a terminal having an always-on function to maintain an IP connection and to receive the VoIP service using the always-on function if the VoIP service over a wireless local area network (LAN) is lost.
2. Discussion of the Background
A wireless communication system has evolved from a system for providing only a simple voice communication service into a system for providing a high-speed data service and a voice service, including VoIP, using packets.
Recently, the current wireless communication system is being replaced with Long Term Evolution (LTE)/System Architecture Evolution (SAE) and LTE Advanced systems for providing services using a packet switching scheme.
When a terminal supporting the LTE/SAE and LTE Advanced networks for providing services using the packet switching scheme is booted up to access the network, a mobile IP is allocated to the terminal in a registration procedure. The allocated mobile IP is maintained to provide an always-on function, which may enable a user to receive a packet service according to the provided service.
In the LTE/SAE and LTE Advanced systems, a voice service is provided in a VoIP scheme using an IP Multimedia Subsystem (IMS), but is not widely used because of load and speed of the network or an expensive fee.
Recently, a VoIP service using a wireless local area network (LAN) module has been provided by including a wireless LAN module, which has a low cost and may include WiFi, WiMAX or Wireless Broadband (WiBro), in a user terminal.
However, since the wireless LAN has a shorter range, it is difficult to guarantee continuity of the VoIP service when a user moves away from an access point (AP).
Conventionally, as shown in FIG. 1, a user terminal UE1 having a wireless LAN module mounted therein accesses a first Proxy-Call Session Control Function (P-CSCF) over a wireless LAN. The user terminal UE1 then performs a registration procedure and repeatedly performs a registration refresh procedure before a session expiration time is reached. However, the user terminal UE1 does not communicate with the P-CSCF during a period between registration refresh procedures (usually, 40 to 60 seconds, not limited thereto). Since the first P-CSCF may not be aware of the state of the user terminal UE1 during this period, the first P-CSCF recognizes the user terminal UE1 to be in a call enable state.
However, if a user terminal UE1 moves beyond the coverage of a wireless LAN and access to an AP is disconnected between registration refresh procedures, the first P-CSCF does not recognize that the connection between the user terminal UE1 and the AP has disconnected until the timing arrives for performing the next registration refresh procedure.
If the connection between the user terminal UE1 and the AP is disconnected between registration refresh procedures, and a counterpart's terminal wishes to perform VoIP communication with the user terminal UE1 and sends a Session Initiation Protocol (SIP) Invite message through a third P-CSCF to try to establish VoIP call connection with the user terminal UE1 during this time, the first P-CSCF sends a response message to the SIP Invite message to the counterpart's terminal and sends the SIP Invite message to the user terminal UE1. However, since the connection between the user terminal and the AP has been disconnected, the first P-CSCF does not receive a response message to the SIP Invite message from the user terminal UE1 within an acknowledgement time. Thus, the SIP request may time out.
The first P-CSCF does not receive the response message from the user terminal UE1, and so may waste resources by retransmitting the SIP Invite message to the user terminal UE1. Once the first P-CSCF recognizes that the user terminal UE1 is in a call disable state, such as if the retransmission time has expired, the first P-CSCF sends a message indicating that the user terminal UE1 does not respond to the counterpart's terminal, terminates the call, and allows the CSCF to delete SIP registration information of the user terminal UE1.
As described above, in the related art, if the connection between the user terminal UE1 and the AP is disconnected, the first P-CSCF performs an unnecessary operation of retransmitting the SIP Invite message to the disconnected user terminal UE1. Also, the counterpart's terminal, which tries to establish VoIP call connection with the user terminal UE1, waits for a ringback tone while the SIP Invite message is retransmitted to the disconnected user terminal UE1.
As shown in FIG. 2, if VoIP communication is initiated between the user terminal UE1 connected to the first P-CSCF and the counterpart's terminal connected to the third P-CSCF over the wireless LAN, the user terminal UE1 repeatedly performs the registration refresh procedure even during VoIP communication. If the connection between the user terminal UE1 and the AP is disconnected during VoIP communication between the user terminal UE1 and the counterpart's terminal, a Real-time Transport Protocol (RTP) session between the user terminal UE1 and a media sever is disconnected, but the first P-CSCF and the third P-CSCF may not recognize that the user terminal UE1 is in a call disable state.
As a result, the counterpart's terminal maintains the VoIP call via the media server during a predetermined time even if the disconnected user terminal UE1 is in the call disable state. Thus, a delay occurs during call.